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[client] audio: open device earlier

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The actual time between opening the device and the device starting to pull
data can range anywhere between nearly instant and hundreds of
milliseconds. To minimise startup latency, open the device as soon as the
first playback data is received from Spice. If the device starts earlier
than required, insert a period of silence at the beginning of playback to
avoid underrunning. If it starts later, just accept the higher latency and
let the adaptive resampling deal with it.
This commit is contained in:
Chris Spencer 2022-02-12 13:34:37 +00:00 committed by Geoffrey McRae
parent 1ca43c4727
commit 70158a64e7
1 changed files with 33 additions and 11 deletions
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44
client/src/audio.c
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@ -37,7 +37,8 @@
typedef enum
{
STREAM_STATE_STOP,
STREAM_STATE_SETUP,
STREAM_STATE_SETUP_SPICE,
STREAM_STATE_SETUP_DEVICE,
STREAM_STATE_RUN,
STREAM_STATE_KEEP_ALIVE
}
@ -99,6 +100,7 @@ typedef struct
int sampleRate;
int stride;
int deviceMaxPeriodFrames;
int deviceTargetStartFrames;
RingBuffer buffer;
RingBuffer deviceTiming;
@ -217,6 +219,19 @@ static int playbackPullFrames(uint8_t * dst, int frames)
if (audio.playback.buffer)
{
if (audio.playback.state == STREAM_STATE_SETUP_DEVICE)
{
// If necessary, slew backwards to play silence until we reach the target
// startup latency. This avoids underrunning the buffer if the audio
// device starts earlier than required
int offset = ringbuffer_getCount(audio.playback.buffer) -
audio.playback.deviceTargetStartFrames;
if (offset < 0)
ringbuffer_consume(audio.playback.buffer, NULL, offset);
audio.playback.state = STREAM_STATE_RUN;
}
// Measure the device clock and post to the Spice thread
if (frames != data->periodFrames)
{
@ -329,7 +344,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.channels = channels;
audio.playback.sampleRate = sampleRate;
audio.playback.stride = channels * sizeof(float);
audio.playback.state = STREAM_STATE_SETUP;
audio.playback.state = STREAM_STATE_SETUP_SPICE;
audio.playback.deviceData.periodFrames = 0;
audio.playback.deviceData.nextPosition = 0;
@ -362,8 +377,6 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
audio.playback.timings = ringbuffer_new(1200, sizeof(float));
audio.playback.graph = app_registerGraph("PLAYBACK",
audio.playback.timings, 0.0f, 200.0f, audioGraphFormatFn);
audio.playback.state = STREAM_STATE_SETUP;
}
void audio_playbackStop(void)
@ -390,8 +403,9 @@ void audio_playbackStop(void)
break;
}
case STREAM_STATE_SETUP:
// We haven't actually started the audio device yet so just clean up
case STREAM_STATE_SETUP_SPICE:
case STREAM_STATE_SETUP_DEVICE:
// Playback hasn't actually started yet so just clean up
playbackStop();
break;
@ -679,7 +693,7 @@ void audio_playbackData(uint8_t * data, size_t size)
spiceData->nextPosition += srcData.output_frames_gen;
}
if (audio.playback.state == STREAM_STATE_SETUP)
if (audio.playback.state == STREAM_STATE_SETUP_SPICE)
{
// In the worst case, the audio device can immediately request two full
// buffers at the beginning of playback. Latency corrections at startup can
@ -688,10 +702,18 @@ void audio_playbackData(uint8_t * data, size_t size)
// before starting playback to minimise the chances of underrunning
int startFrames =
spiceData->periodFrames * 2 + audio.playback.deviceMaxPeriodFrames * 2;
if (spiceData->nextPosition >= startFrames) {
audio.audioDev->playback.start();
audio.playback.state = STREAM_STATE_RUN;
}
audio.playback.deviceTargetStartFrames = startFrames;
// The actual time between opening the device and the device starting to
// pull data can range anywhere between nearly instant and hundreds of
// milliseconds. To minimise startup latency, we open the device
// immediately. If the device starts earlier than required (as per the
// `startFrames` value we just calculated), then a period of silence will be
// inserted at the beginning of playback to avoid underrunning. If it starts
// later, then we just accept the higher latency and let the adaptive
// resampling deal with it
audio.playback.state = STREAM_STATE_SETUP_DEVICE;
audio.audioDev->playback.start();
}
double latencyFrames = actualOffset;